This change improves echo canceller transparency by enabling the use
of a non-capped ERLE when computing the residual echo spectrum for
dominant nearend detection.
Experimentation has shown that the feature improves echo canceller
transparency and user ratings.
Implementation CL:
https://webrtc-review.googlesource.com/c/src/+/221920
Bug: webrtc:12870
Change-Id: I7dc66810e8300cd35321bcd5b9fae9bc3386836d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234841
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35186}
`a && b` only evaluates b if a is true. `a & b` always evaluates
both a and b. If a and b are of type bool, `&&` is usually what you
want, so clang now warns on `&` when both arguments are of type bool.
In the one case where this fires in webrtc, it isn't important if we
evaluate both branches, so I went with `&&`.
Bug: chromium:1255745
Change-Id: I7fd215778fca62e0d5ca64ab0cf1142942eb7304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234600
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35181}
Sends a VideoLayersAllocation header extension if frame rate change more than 5fps since the last time it was sent with valid frame rate and resolution.
Bug: webrtc:12000
Change-Id: I2572c966025cc2c22743bbe2187cec7cceb86d01
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234752
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35180}
Encoder info will be modified at runtime. In fact, we should reduce the
number of 'full' ReconfigureEncoder(). If only need subset of it at
runtime, consider handle it in VideoStreamEncoder::EncodeVideoFrame().
Consider two cases:
Re-configure scalers when encoder info changed. Consider two cases:
1. When the status of the scaler changes from enabled to disabled, if we
don't do this CL, scaler will adapt up/down to trigger an unnecessary
full ReconfigureEncoder() when the scaler should be banned.
2. When the status of the scaler changes from disabled to enabled, if we
don't do this CL, scaler will not work until some code trigger
ReconfigureEncoder(). In extreme cases, the scaler doesn't even work for
a long time when we expect that the scaler should work.
This CL aims to make scalers work properly when encoder info changed.
BUG: None
Change-Id: Iec17730b5fac5e642c0fb2d9b11c5b7434f0a220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233384
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35175}
Internal refactoring of AGC2. This CL is needed in preparation for its
child CL to correctly show the upcoming changes in the diff.
Bug: webrtc:7494
Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35170}
It seems the Android CTS tests only verify that 16x16 aligned resolutions
are supported.
This change checks the validity of input frame's size when initialing
or encoding processes are about to start using H/W MediaCodec.
This change has additional APIs to retrieve
|requested_resolution_alignment| and |apply_alignment_to_all_simulcast_layers|
from JAVA VideoEncoder class and its inherited classes. HardwareVideoEncoder
using MediaCodec has values of 16 and true for above variables.
Bug: webrtc:13089
Change-Id: I0c4ebf94eb36da29c2e384a3edf85b82e779b7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229460
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35169}
This is a follow-up to change 232904 that also validates that the
timestamp from the heartbeat ack isn't negative (which the fuzzer
managed to set it to).
Bug: chromium:1252515
Change-Id: Idaac570589dbdaaee67b7785f6232b60226e88e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234582
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35168}
The advantage is that GCD maintains the internal thread pool and
spawns threads when needed. I would expect the behavior to be
almost identical to creating a thread using PlatformThread.
Bug: webrtc:13237
Change-Id: Ie4406b5d128c244f66a73830d5a27f2d8fd88549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35165}
In case if primary encoder can't be instantiated (max number of
instances has reached, for example), use fallback encoder.
Bug: none
Change-Id: I477bdeb7af4dcce50e36b1804ffc6ad2ab004dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234500
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35161}
This avoids an additional step where we originally copied content from
PipeWire buffer to a temporary location and from there to DesktopFrame.
This results into less copy operations and hopefully to faster
screensharing.
I didn't do some exact measures, but simply running htop while sharing a
4k screen I can see following results (usage per top 5 processes):
1) Without this change - 66%, 64%, 26% 23%, 10%
2) With this change - 41%, 39%, 19%, 17%, 12%,
Bug: webrtc:13239
Change-Id: I6a661ecc96bfeef370c1a5a3b9dc5e3c0fc665c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231684
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35156}
This is a reland of f2177f6612079ccce9c320ea7e77bc934c684f5c
Original change's description:
> PipeWire capturer: implement proper DMA-BUFs support
>
> Currently both KWin (KDE) and Mutter (GNOME) window managers don't
> use DMA-BUFs by default, but only when client asks specifically for
> them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
> While current implementation works just fine on integrated graphics
> cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
> code either crashes or screensharing is slow and unusable.
>
> To fix this, DMA-BUFs has to be opened using OpenGL context and not
> being directly mmaped(). This implementation requires to use DMA-BUF
> modifiers, as they are now mandatory for DMA-BUFs usage.
>
> Documentation for this behavior can be found here:
> https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox
>
> Bug: chromium:1233417
> Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34889}
Bug: chromium:1233417
Change-Id: I308501d86ec18ab6df9bcee569c4b72df7926549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35152}
When new packets are enqueued after a dead-period where media debt is
zero, that time slice should not be used to reduce the debt for the
new packet.
Bug: webrtc:10809
Change-Id: Ifb960548e6aa349b79f37743cbfed78a5c937a13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234081
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35143}
This CL does *not* change the behavior of the AGC2 adaptive digital
controller - bitexactness verified with audioproc_f on a collection of
AEC dumps and Wav files (42 recordings in total).
Tested: compiled Chrome with this patch and made an appr.tc test call
Bug: webrtc:7494
Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35140}
